Re: [rtcweb] SIP MUST NOT be used in browser?[was RE: Remoterecording - RTC-Web client acting as SIPREC session recordingclient]

Igor Faynberg <> Thu, 08 September 2011 13:53 UTC

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Date: Thu, 08 Sep 2011 09:55:01 -0400
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Subject: Re: [rtcweb] SIP MUST NOT be used in browser?[was RE: Remoterecording - RTC-Web client acting as SIPREC session recordingclient]
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Finally, I am getting an answer to my question!  Thanks, Silvia!

One question: Is it possible to get the full library source code?  (I 
cannot find the code, but this is the one that sends the 
INVITE as far as I figured out.  Most important, I cannot find how 
notifications are handled.)



On 9/7/2011 9:15 PM, Silvia Pfeiffer wrote:
> If implementations count for anything, check out:
> and
> They use SIP with web sockets.
> Cheers,
> Silvia.
> On Thu, Sep 8, 2011 at 8:30 AM, Matthew Kaufman
> <>  wrote:
>> On 9/7/11 12:20 PM, Randell Jesup wrote:
>>> I also started from the same point - assume SIP.  SIP gives you all the
>>> things that the zillions of hours and emails to define it and define
>>> extensions and secure it provides, without having to reinvent all those
>>> wheels (or ask app developers to reinvent them).  Why go through the
>>> horrible pain of choosing something else, or why throw the app developers to
>>> the wolves to fend for themselves?
>>> However...
>>> Two things have swayed me.  The primary one is the suggestion of
>>> Offer/Answer in the browser.  This breaks out the important negotiation
>>> piece that almost any application would need, and while not perfect, SDP O/A
>>> is a zillion times simpler than SIP with all the extensions one could use.
>> I agree with this. While I am also opposed to SDP O/A, these are two
>> unrelated arguments to have... and not baking a SIP phone into the browser
>> is *more* important than avoiding a repeat of the offer/answer problems.
>>> The other thing that swayed me was thinking about federation and the apps
>>> that will be built with this.  A webrtc app talks to its (web)server, other
>>> webrtc clients running the app that talk to the server, and to other webrtc
>>> applications/networks that federate with it (and their clients).
>>> Federation is in the same hands as the person who provides/wrote the app.
>>>   If they have no interest in federation you can't force it, and they may
>>> have no use for all the fancy SIP standards.
>> And for numerous types of apps (think: server-based augmented reality
>> systems), "federation" doesn't even make sense.
>>> On the other hand, if they *want* to either provide access to the wider
>>> communication net that is the PSTN network, now or in the future, or they
>>> want easy federation with other networks, it behooves them to use SIP or
>>> something very close to it or equivalent/convertible (at a basic level at
>>> least) to it.
>>> So what conclusions do I draw from this?
>>> 1) O/A via SDP in the browser simplifies a lot of things (including
>>> handling new codecs, etc).  It doesn't extremely limit an application,
>>> though we should think about how an application can interact with the
>>> fmtp/etc parameters used.
>> I agree that it would simplify some interop cases, but at an unfortunate
>> cost in lack of flexibility and functionality. Still not nearly as bad as if
>> we put a full SIP stack in there though.
>>> 2) SIP as a *separate* item that can be cleanly and easily *added* to a
>>> webrtc app to handle the call setup/etc is a good idea.
>> I would be open to looking at this again, *after* RTC is already in browsers
>> and successful, to see if it actually solves a real use case.
>> Matthew Kaufman
>> _______________________________________________
>> rtcweb mailing list
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